The present disclosure herein relates to a data transmission system, an encoding apparatus and an encoding method, and more particularly, to a data transmission system, an encoding apparatus and an encoding method that encode and transmit original packets in units of generation.
In the field of voice over internet protocol (VoIP) related to an IP based voice communication technique, packet loss significantly affects the quality of voice call. In order to decrease influence on the packet loss in the field of VoIP, a receiver side may recover a lost packet by using a forward error correction (FEC) technique generally. It is possible to apply, to the VoIP, a random linear coding (RLC) technique as another loss recovery technique different from the FEC, in which case the original packets that are generated by a VoIP codec are logically split by a transmitter in units of generation, and original packets that belong to the same generation are encoded by the RLC. In this case, the size of all generations (the number of original packets that belong to a generation) is constantly maintained. For example, Korean Patent Publication No. 10-2010-0081900 (published on Jul. 15, 2010) discloses “DATA TRANSMISSION AND RECEPTION METHOD USING RANDOM LINEAR CODING”. In such a typical RLC method, encoding/decoding may be performed only when a predetermined number of packets all gather. That is, since a plurality of original packets gathers to make up a single generation and the generation becomes a unit for encoding/decoding, encoding/decoding may be performed only when all original packets that belong to a single generation gather. Thus, it is inevitable that the transmitter and the receiver have further delay times, respectively. For example, in the situation in which like VoIP traffic, packets are sequentially generated at a time interval, ten packets should gather for encoding when the size of the generation is ten, thus a further delay time occurs for which no information is transmitted to the receiver side until a tenth packet is generated after a first packet is generated. Thus, VoIP traffic has a limitation in that voice call quality decreases by the packet loss and the delay time.